Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time data over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be extended to deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Rockwall.
SIP is usually associated with a way to emulate the functionality of ordinary telephony over an IP network. It is replacing the older, less versatile protocols used in the past such as H.323 and MGCP. These earlier protocols performed at a very low level to connect IP phones to the public telephone network. SIP, on the other hand, provides an elegant and well-defined way to network the enterprise. For instance, SIP uses email addresses as the SIP address instead of a telephone number over the standard phone network.
SIP trunking provides multiple advantages to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the high subscription costs of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by delivering both voice and data over the very same connection. Businesses will have the manageability to route calls to preferred carriers and the redundancy of using several service providers.
